While analyzing the RTP stream, I noticed this phenomenon on the FreePBX. There are sequences with error incorrect timestamps then there is an increase in jitter plus error wrong sequence number, and then the jitter turns to normal.
This only happens during IVRs.
Call Flow:
SIP Provider → FreePBX → Other Call Center Server (IVR is played here)
It is likely to be a design issue with your version of Asterisk, if the timestamps are jumping, when the media source changes, without a change in SSRC, but I don’t know if it has been fixed, but I’d expect you will have to change the Asterisk version.